webrtcbin test: Add test for various cases where get_request_pad is meant to fail

This should ensure that the recently added code works.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-03-29 19:47:21 -04:00
parent 7f29486ba4
commit 913d308e22

View file

@ -3364,6 +3364,98 @@ GST_START_TEST (test_bundle_codec_preferences_rtx_no_duplicate_payloads)
GST_END_TEST;
GST_START_TEST (test_reject_request_pad)
{
struct test_webrtc *t = test_webrtc_new ();
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiver *trans, *trans2;
guint offer_media_format_count[] = { 1, };
guint answer_media_format_count[] = { 1, };
VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL, NULL);
VAL_SDP_INIT (offer_media_formats, on_sdp_media_count_formats,
offer_media_format_count, &payloads);
VAL_SDP_INIT (answer_media_formats, on_sdp_media_count_formats,
answer_media_format_count, &payloads);
const gchar *expected_offer_setup[] = { "actpass", };
VAL_SDP_INIT (offer_setup, on_sdp_media_setup, expected_offer_setup,
&offer_media_formats);
const gchar *expected_answer_setup[] = { "active", };
VAL_SDP_INIT (answer_setup, on_sdp_media_setup, expected_answer_setup,
&answer_media_formats);
const gchar *expected_offer_direction[] = { "recvonly", };
VAL_SDP_INIT (offer, on_sdp_media_direction, expected_offer_direction,
&offer_setup);
const gchar *expected_answer_direction[] = { "sendonly", };
VAL_SDP_INIT (answer, on_sdp_media_direction, expected_answer_direction,
&answer_setup);
GstCaps *caps;
GstHarness *h;
GstPad *pad;
GstPadTemplate *templ;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_pad_added = _pad_added_fakesink;
gst_util_set_object_arg (G_OBJECT (t->webrtc1), "bundle-policy",
"max-bundle");
gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
"max-bundle");
/* setup recvonly transceiver */
caps = gst_caps_from_string (VP8_RTP_CAPS (96));
direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
g_signal_emit_by_name (t->webrtc1, "add-transceiver", direction, caps,
&trans);
gst_caps_unref (caps);
fail_unless (trans != NULL);
h = gst_harness_new_with_element (t->webrtc2, "sink_0", NULL);
add_fake_video_src_harness (h, 96);
t->harnesses = g_list_prepend (t->harnesses, h);
test_validate_sdp (t, &offer, &answer);
/* This should fail because the direction is wrong */
pad = gst_element_get_request_pad (t->webrtc1, "sink_0");
fail_unless (pad == NULL);
g_object_set (trans, "direction",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, NULL);
templ = gst_element_get_pad_template (t->webrtc1, "sink_%u");
fail_unless (templ != NULL);
/* This should fail because the caps are wrong */
caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
pad = gst_element_request_pad (t->webrtc1, templ, "sink_0", caps);
fail_unless (pad == NULL);
gst_caps_unref (trans->codec_preferences);
trans->codec_preferences = NULL;
/* This should fail because the kind doesn't match */
pad = gst_element_request_pad (t->webrtc1, templ, "sink_0", caps);
fail_unless (pad == NULL);
gst_caps_unref (caps);
/* This should succeed and give us sink_0 */
pad = gst_element_get_request_pad (t->webrtc1, "sink_0");
fail_unless (pad != NULL);
g_object_get (pad, "transceiver", &trans2, NULL);
fail_unless (trans == trans2);
gst_object_unref (pad);
gst_object_unref (trans);
gst_object_unref (trans2);
test_webrtc_free (t);
}
GST_END_TEST;
static Suite *
webrtcbin_suite (void)
{
@ -3407,6 +3499,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_renego_lose_media_fails);
tcase_add_test (tc,
test_bundle_codec_preferences_rtx_no_duplicate_payloads);
tcase_add_test (tc, test_reject_request_pad);
if (sctpenc && sctpdec) {
tcase_add_test (tc, test_data_channel_create);
tcase_add_test (tc, test_data_channel_remote_notify);